Abstract

Transmitting real-time audio or video applications over the Internet is a challenge for current networking technology. Reduced overhead and enhancement of services are the motivations for deploying voice communications. The integration of voice, video, and data encounters a variable amount of jitter and delay. Typically, packet loss ranges from 0% to 20% and one-way delay ranges from 5 to 500 msec. Reducing jitter delay involves buffering of audio packets at the receiver so that the slower packets arrive sequentially on time at the destination. Adaptive jitter buffering at the receiver improves the quality of voice connections on the Internet. In this paper, the existing jitter buffer model was further enhanced by proposing a model to change the audio codecs dynamically. The audio codecs are changed from a higher bit rate to a lower bit rate during an established call session, reducing the packet loss and improving the call performance.

Description

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